Fiter design is the process of designing a filter.

learn more… | top users | synonyms

0
votes
0answers
9 views

Differentials - Differences: Non causality in the system

I'm still learning DSP and referring to Oppenheim video lectures. In that lectures, differential difference equation is obtained for IIR filter design, in Lecture 14. $$\mathcal{L}[\frac{\mathrm ...
0
votes
0answers
24 views

Which method to choose to get linear phase response zero group delay low-pass filter?

Background: I do time-dependent simulations of physical systems. Like this: http://journals.aps.org/prb/abstract/10.1103/PhysRevB.91.115431 During the simulation, observables are recorded and later ...
1
vote
1answer
30 views

Multiple parallel IIR bandpass filters with different centre frequencies combined into one filter for optimisation… possible?

Before I begin, I have read these already: IIR filter parallelization Concept of combining multiple FIR Filters into 1 FIR filter I am processing audio using Csound in a high performance real time ...
1
vote
1answer
22 views

Transform matrix creation in compressed sensing

In Compressed sensing ECG reconstruction I would like to use bi orthogonal wavelet basis as transform/sparsifying domain. I found out that ' wfilters ' command in matlab will give 4 type of filters, ...
0
votes
0answers
16 views

Specifications (N) FIR Low Pass Resamplig, Hamming Window

I'm learning about DSP and I've doubts. The resampling process can be do it with only one filter. $L$: Increment Sample Rate Integer $M$: Decrement Sample Rate Integer $f_S$: Frequency Sampling ...
0
votes
2answers
108 views
+50

Problem designing a specific filter

I have the next problem. $H_{c1}(j\omega )$ is the ideal antialising filter and $H_{c2}(j\omega )$ is a real one. I'm asked to design $H(e^{j\Omega })$ so that $y[n]$ in the second diagram (the one ...
0
votes
1answer
30 views

Short lowpass filter that also extrapolates lines

I'm looking for a short lowpass filter that also extrapolates lines, [0 1 2 3] --> 4, so I need (correct me) $H([\ 0\ \ 0.25\ \ 0.5\ ]) \approx [ 1\ \ 2\ \ 0 ]$ . How close can one get to this with 2 ...
0
votes
1answer
25 views

MATLAB fvtool: Setting the passband ripple below 0dB for a FIR Filter

I'm new to using filter tools in MATLAB and I'm having a bit of difficulty. I'm designing a lowpass FIR filter and I'm trying to keep the passband ripple below 0dB in the magnitude response. So far, ...
0
votes
0answers
27 views

How to add single wavelength discrimination to the Kutulakos & O'Toole paper on a generalized Light Transport Matrix system

I wish to add single wavelength discrimination to the excellent paper Transport-Aware Imaging, Proceedings of SPIE Vol 9376, 937606 @ 2015 SPIE ,authored by Professor Kiriakos Kutulakos and Ph.D ...
0
votes
3answers
72 views

3dB and 6dB Per Octave Rolloff Digital Filters / Generating Pink and Brown Noise

So I would like to have a design method for filters with 3dB and 6dB per octave roll off -- for generating pink and brown noise respectively. I know the following 'pinking' filters exist: Filter to ...
0
votes
2answers
28 views

Interpolated impulse response for fraction delay?

I need to find a way to create some fractional delay in a signal processing application that I am working on. Separately to this, I have been messing around with basic filter design (despite reading ...
3
votes
1answer
44 views

Problem designing a specific FIR filter

Consider an LTI system whose impulse response is $$h[n]=\frac{1}{2^n}u[n]+\frac{1}{3^n}u[n]$$ The input signal to this system is $x[n]$ and is null for $n<0$ but may or may not be null for $n=0$. ...
4
votes
1answer
80 views

Are there other filter models besides IIR and FIR for processing digital signals?

I am very familiar with IIR and FIR filters for processing digital signals (audio is my main interest). I am keen to know if there are alternative filters based on different mathematical models to IIR ...
0
votes
0answers
17 views

Optimize an impulse response function to reduce PAPR

I would like to reduce PAPR (Peak to average power ratio) for single carrier system by decreasing magnitude of the side lobes of impulse response function (Nyquist pulse) in time domain at transmitter ...
0
votes
0answers
18 views

Problem with DFT Filter Banks and MATLAB

I'm posting the results of dft FB on matlab, and I would discuss about them. To generate the analysis and synthesis prototype i used a simple method for pseudo-QMF, published by Mitra in his ...
0
votes
1answer
38 views

Is there a set formula for calculating FIR coefficients (for a LPF ir 96 taps)?

I need to calculate a series of coefficients for several FIR filters having 96 taps is there a standard or set formula for calculating these, it is a low pass filter.
0
votes
1answer
58 views

how to design LPH and HPF from APF

I came across a concept that says we can design LPH and HPF from APF when the transfer function of the two APFs given as $$\begin{align}A_0(z)&=1\\ A_1(z)&=\frac{-a+z^{-1}}{1-az^{-1}}\\\\ ...
0
votes
1answer
31 views

how to find breakpoints of a signal based on the frequency of the signal

There are 4 cos signals with 10,20,30 and 100 Hz each. The main signal is the sum of those 4 and I have to create a low pass filter, on the main signal for the frequency's > 50 Hz. ...
0
votes
1answer
49 views

How Matched filter is transformed due to frequency modulation?

Intro: Consider a signal source S which emits a PAM modulated signal, bits encoded as +g(t) and ...
0
votes
0answers
26 views

Smoothing the discrete acceleration

In order to know if my signal is increasing or decreasing, I'm using the discrete derivative $y[n] = x[n] - x[n-1]$ or a smoothed version of it (for example Exponential Weight Moving Average of $y[n]$ ...
2
votes
0answers
34 views

Turn a decision problem into a linear filter problem

Let $x[n]$ be a time-series, and two filters: $A[n] = a_0 x[n] + a_1 x[n-1] + ... a_q x[n-q]$ $B[n] = b_0 x[n] + b_1 x[n-1] + ... b_r x[n-r]$ I think the answer is No, but is there a way to have ...
0
votes
5answers
267 views

3dB-Cut off frequency of moving average

I've read the thread: Cut off frequency of moving average And I use the second answer in my algorithm to calculate the 3dB cut off frequency of my filter, which works great, as my filter length is ...
1
vote
2answers
47 views

Complex band pass filter

I'm trying to find some mathematical docs about the complex (analytic signal) band pass filtering in discrete time. I've read some text already but most of them describes the problem in real time ...
1
vote
5answers
123 views

What kind of filter would have a sawtooth frequency response?

I want a complex FIR or IIR filter whose response is approximately a sawtooth in the frequency domain. How should this be formally described, and what formula or algorithm can I use to calculate taps ...
1
vote
2answers
51 views

Design one of 4 type of FIR filter by window function?

There're 4 type of FIR to chose from symetry ,even symetry ,odd antisymetry, even antysymetry, odd Also, there are many type of window like rectangular, Hanning, Kaiser,etc So how to use a window ...
1
vote
2answers
98 views

What does this signal filter do? (coming from finance)

In finance, many "indicators" are used in order to make prediction about if a price has an increasing or decreasing trend over time. Among them, MACD is a popular one. There are many slighlty ...
0
votes
2answers
68 views

Filter Design for Phase Response

How can I design an all pass filter to have a constant phase shift over a bandwidth centered around a carrier? I don't care about phase shift outside the band. I would like to have the filter in ...
0
votes
2answers
67 views

Is the below filter linear phase

$$h(t) = \frac{1}{1 + t^2}$$ and is it IIR or FIR filter. I tried finding the Laplace transform of this filter to get the data flow diagram with 5 taps and T=2s, however, I am unable to solve this. ...
0
votes
2answers
156 views

High-pass filtering pre-processing before computing audio features

I want to compute acoustic features on a set of audio files. These features can be classical audio features as spectral flux, but also acoustic indices as proposed in the seewave R package ). I ...
0
votes
0answers
16 views

Denoising of Images capture by digital camera using Butterworth Filter

Right now, i am doing a research on how the Butterworth filter can reduce the noise in images captured by digital camera. For start, what i need is images which contain noise in it. My question is; ...
1
vote
1answer
39 views

How to remove or filter the drift problem in measured Strain signal?

I have the strain signal of a lateral beam of a car measured at sampling rate 1,200Hz from data acquiring system. Even after using temperature compensation in strain gage, we are getting drift. So I ...
1
vote
1answer
55 views

How to eliminate noise that has the same frequency of my signal

I have an audio signal , and I want to remove noise from it. I have applied a bandpass filter in order to remove frequencies that are not part of mi signal. But if there is background noise (some ...
0
votes
1answer
36 views

Design a FIR filter with a Kaiser window

Here is the exercise I am working on. Use the window design method with the Kaiser window to design a linear phase high pass FIR filter of minimum order that meets the folowing specification: ...
1
vote
1answer
65 views

IIR Filter Design Issues : MATLAB

I am designing an IIR Filter with the following designs constraints: Passband $(0.75\pi,\pi)$ Stopband $(0,0.70\pi)$ No more than $0.5$ % ripple in the passband No less than $40$ dB attenuation in ...
0
votes
0answers
44 views

Relation of roots of a polynomial to amplitude distribution of its coefficients

Consider $g(x)=\sum_{k=0}^{p}a_kx^{k}$, and let $\{\tau_1,\tau_2,\cdots,\tau_p\}\in\mathbb{C}$ be the roots of $g(x)$. What is the relation between how $\{a_0,a_1,\cdots,a_p\}$ are distributed (in ...
1
vote
0answers
36 views

Time-domain filtering to match a target time-varying frequency response

I would like to implement a filter in the time domain based on a desired frequency response that changes over time. The initial goal is a bandpass filter with a time-varying center frequency, where ...
0
votes
1answer
95 views

Why lattice-structured all-pole IIR filter becomes stable, if the k-coefficients satisfies $|k_i| < 1$? [closed]

Why the lattice-structured all-pole IIR filter becomes stable, if the k-coefficients satisfies $|k_i| < 1$? Anyone who can tell me the way how to prove it, would be so thankful for me :) ps. If ...
0
votes
1answer
64 views

How to find out the farrow coefficients if FIR coefficients are given?

FIR filter coefficients are known. Then what is the matlab code or function that is used to determine the corresponding farrow structure coefficients?
3
votes
1answer
53 views

Discrete-time Biquad filter: Relation between peak frequency and pole frequency

This question arose from a discussion in the comments to this question and its answer. The author of that question discussed a discrete-time second order filter described by the following difference ...
1
vote
1answer
31 views

dealing with Huge Length (Order) FIR filters

I'm studying a project. My signal is audio 8KHz Mono, 8 bits in Low frequencies 70Hz until 1200Hz. When I perform the calculus with the transition band is very narrow (Transition Frequency) my Order ...
0
votes
0answers
21 views

comb filter design in wireless communication

How can I implement comb filter in reducing noise in wireless communication? I am new in signal processing and right now I am still learning about the comb filter. Can I use it to reduce/filter noise ...
1
vote
1answer
54 views

Special case peak/notch filter for exactly 1/32 sampling rate?

I need a filter at 1/32 my sampling rate. Because it's 1/2n, I was thinking there might be a special case filter for this case. Anything like this?
0
votes
1answer
57 views

Given an FIR filter coefficients, how can I find coefficients of its equalizer filter?

I have an N-tap direct form complex FIR filter with known coefficients [c0 c1 .. cn]. I want to find the coefficients of its equalizer filter which when cascaded ...
0
votes
1answer
16 views

Why ignores constant when applying integral into differential-equation.?

I studied how to draw 'Direct-form I' and 'Direct-form II' from differential equation. The book taught me like this: $$\mbox{Give equation: }{a_0}{y(t)}+{a_1}{y'(t)}+{a_2}{y''(t)} = ...
0
votes
0answers
23 views

Convert simple CT filter to DT with impulse invariance and bilinear transform

Can someone verify my work. Problem Question: Convert simple CT filter to DT with (A) impulse invariance and (B) bilinear transform $$ H_c(j \Omega) = \begin{cases} |\Omega|, & |\Omega| ...
3
votes
2answers
574 views

What does 'canonical' mean?

What does 'canonical' mean? e.g. 'write IIR or FIR in canonical form' I get the gist of it but maybe someone here has a precise definition. In a non-DSP context it would mean something like ...
1
vote
1answer
30 views

Multiple cheap IMUs vs one expensive IMUs

In my project, there is a possibility of either buying one expensive IMU or 5 cheaper IMUs. They have to be cheaper due to budget limitations, and the idea is to use them in order to be able to do ...
2
votes
1answer
65 views

Practical vs ideal lowpass interpolator

Consider a signal with a sample rate $f_s = 44.1$ kHz. Let us upsample the signal by a factor of $L = 2$ and interpolate the zeros. An ideal lowpass interpolator would have a gain of $L$ and a cutoff ...
0
votes
4answers
109 views

Design Hilbert filter at matlab (pure 90 phase and magnitude 0db)

I need to design a filter able to take a measurement of alternate current and shift the phase of the input by 90°. The filter should work between 40 and 60Hz and at this range should have unity ...
0
votes
3answers
93 views

Designing a 2x interpolation filter for audio

I want to perform 2x interpolation on an audio signal sampled at 44.1KHz by upsampling the signal by adding a zero after each original sample and then using a lowpass filter to interpolate the ...